- Use frame_num instead of pic_num to set the long_term_pic_num
fixing 10 interlaced tests in fluster test suite: JVT-AVC_V1
- Send the slice offset only once in case of interlaced content.
Fixing 5 interlaced tests in fluster test suite: JVT-AVC_V1.
- The default value for top and bottom field flag should be 0 in the
case of a progressive content.
- Use short and long term refs helper getter method to retrieve the
reference frames according its none existing and interlaced state
- Reorganize the find_next_slot_idx code to be easier to read.
Co-authored-by: Daniel Almeida <daniel.almeida@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7854>
Extended is identical to main but allows FMO/ASO features to be used,
and prevent using CABAC.
Using similar logic to "baseline", assume that if we support main,
we can also do extended.
This fixes the following fluster vectors, which otherwise would fail when trying to link the parsebin pad.
BA3_SVA_C
MR6_BT_B
MR7_BT_B
MR8_BT_B
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7854>
Before this patch, there could be duplicate payload types in offers that
have, within a media section, multiple codecs and RTX enabled:
```
m=video 9 UDP/TLS/RTP/SAVPF 96 97 97 <-- HAS DUPLICATES
a=sendrecv
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 transport-cc
a=rtpmap:97 H264/90000
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=rtcp-fb:97 transport-cc
a=rtpmap:97 rtx/90000 <--------- PT IS DUPLICATE
a=fmtp:97 apt=96
```
Fix this by populating the media_mapping array with all media formats
rather than only the first one. The added test case reproduces the issue,
which fails without this patch.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8259>
Adds a new signal to webrtcbin, to allow for placement
of an object after rtp, before sendbin. This is usable for
objects such as congestion control elements, that don't want
to be burdened by the synchronization requirements of rtpsession.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7940>
The gst_srtp_dec_decode_buffer() function modifies the input buffer after making
it writable, so the pointer might change as well, depending on the refcount of
the buffer.
This issue was detected using a netsim element upstream of the decoder in a
WebRTC pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8198>
We can use gst_uri_from_string_with_base () to join base url
and the fragment url path.
The previous method of forming base url in update_base_url(),
by looking for the string 'manifest' or 'Manifest' is insufficient.
A query may include these string in their paths and thus an invalid
base url string will be kept.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8193>
Use GST_VIDEO_DECODER_ERROR instead of just erroring out
unconditionally, so that the error handling behaviour is
determined by the "max-errors" property and we'll just
continue after decoding errors now instead of erroring out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8163>
Parsing the whole caps as SDP media only to retrieve the fmtp field afterwards
seems a bit superfluous. By looking up the a-fmtp attribute directly the number
of allocations in this function gets down a bit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8125>
There was different behaviour if the proxy was configured through
properties or environment. For properties libcurl would be configured
with any auth, but for environment libcurl would default to using basic.
Now any auth is set for both configuration methods.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7935>
